US5787391A - Speech coding by code-edited linear prediction - Google Patents

Speech coding by code-edited linear prediction Download PDF

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US5787391A
US5787391A US08/658,303 US65830396A US5787391A US 5787391 A US5787391 A US 5787391A US 65830396 A US65830396 A US 65830396A US 5787391 A US5787391 A US 5787391A
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vector
gain
multiplying
selecting
pitch period
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Takehiro Moriya
Akitoshi Kataoka
Kazunori Mano
Satoshi Miki
Hitoshi Omuro
Shinji Hayashi
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Nippon Telegraph and Telephone Corp
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Nippon Telegraph and Telephone Corp
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Priority claimed from JP07053493A external-priority patent/JP3148778B2/ja
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/005—Correction of errors induced by the transmission channel, if related to the coding algorithm
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/06—Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/06—Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
    • G10L19/07—Line spectrum pair [LSP] vocoders
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/083—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being an excitation gain
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/12—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/12—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
    • G10L19/135—Vector sum excited linear prediction [VSELP]
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L2019/0001—Codebooks
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L2019/0001—Codebooks
    • G10L2019/0002—Codebook adaptations
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L2019/0001—Codebooks
    • G10L2019/0003—Backward prediction of gain
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L2019/0001—Codebooks
    • G10L2019/0004—Design or structure of the codebook
    • G10L2019/0005—Multi-stage vector quantisation

Definitions

  • the present invention relates to a speech coding method, and an apparatus for the same, for performing high efficiency speech coding for use in digital cellular telephone systems. More concretely, the present invention relates to a parameter coding method, and an apparatus for the same, for encoding various types of parameters such as spectral envelope information and power information, which are to be used in the aforementioned speech coding method and apparatus for the same; the present invention further relates to a multistage vector quantization method, and an apparatus for the same, for performing multistage vector quantization for use in the aforementioned speech coding process and apparatus for the same.
  • code-excited linear prediction coding CELP
  • VSELP vector sum excited linear prediction coding
  • multi-pulse coding CELP
  • CELP code-excited linear prediction coding
  • VSELP vector sum excited linear prediction coding
  • multi-pulse coding multi-pulse coding
  • FIG. 15 is a block diagram showing a constructional example of a speech coding apparatus utilizing a conventional CELP coding method.
  • the analog speech signal is sampled at a sampling frequency of 8 kHz, and the generated input speech data is inputted from an input terminal 1.
  • LPC linear prediction coding
  • a plurality of input speech data samples inputted from the input terminal 1 are grouped as one frame in one vector (hereafter referred to as "an input speech vector"), and linear prediction analysis is performed for this input speech vector, and LPC coefficients are then calculated.
  • LPC coefficient quantizing portion 4 the LPC coefficients are quantized, and the LPC coefficients of a synthesis filter 3 possessing the transfer function ⁇ 1/A(z) ⁇ is then set.
  • An adaptive codebook 5 is formed in a manner such that a plurality of pitch period vectors, corresponding to pitch periods of the voiced intervals in the speech, are stored.
  • a gain portion 6 a gain set by a distortion power calculating portion 13 explained hereafter is multiplied by the pitch period vector, which is selected and outputted from the adaptive codebook 5 by the distortion power calculating portion 13 and is then outputted from the gain portion 6.
  • a plurality of noise waveform vectors (e.g., random vectors) corresponding to the unvoiced intervals in the speech are previously stored in a random codebook 7.
  • the gain set by distortion power calculating portion 13 is multiplied by the noise waveform vector, which is selected and outputted from the random codebook 7 by the distortion power calculating portion 13, and outputted from gain portion 8.
  • the output vector of the gain portion 6 and the output vector of the gain portion 8 are added, and the output vector of the adder 9 is then supplied to the synthesis filter 3 as an excitation vector.
  • the speech vector hereafter referred to as "the synthetic speech vector" is synthesized based on the set LPC coefficient.
  • a power quantizing portion 10 the power of the input speech vector is first calculated, and this power is then quantized. In this manner, using the quantized power of the input speech vector, the input speech vector and the pitch period vector are normalized. In a subtracter 11, the synthetic speech vector is subtracted from the normalized input speech vector outputted from the power quantizing portion 10, and the distortion data is calculated.
  • the distortion data is weighted in a perceptual weighting filter 12 according to the coefficients corresponding to the perceptual characteristics of humans.
  • the aforementioned perceptual weighting filter 12 utilizes a masking effect of the perceptual characteristics of humans, and reduces the auditory senses of quantized random noise in the formant region of the speech data.
  • a distortion power calculating portion 13 calculates the power of the distortion data outputted from the perceptual weighting filter 12, selects the pitch period vector and the noise waveform vector, which will minimize the power of the distortion data, from the adaptive codebook 5 and the random codebook 7, respectively, and sets the gains in each of the gain portions 6 and 8. In this manner, the information (codes) and gains selected according to the LPC coefficients, power of the input speech vector, the pitch period vector and the noise waveform vector, are converted into codes of bit series, outputted, and then transmitted.
  • FIG. 16 is a block diagram showing a constructional example of a speech coding apparatus utilizing a conventional VSELP coding method.
  • components which correspond to those shown in FIG. 15, will retain the original identifying numeral, and their description will not herein be repeated.
  • the construction of this speech coding apparatus utilizing the VSELP coding method is similar overall to that of the aforementioned speech coding apparatus utilizing the CELP coding method.
  • the VSELP coding method in order to raise the quantization efficiency, utilizes a vector quantization method which simultaneously determines the gains to be multiplied with the selected pitch period vector and noise waveform vector respectively, and sets them into gain portions 15a and 15b of a gainer 15.
  • CELP Code-Excited Linear Prediction
  • VSELP Vector Sum Excited Linear Prediction
  • a low-delay code excited linear prediction (LD-CELP) coding method is a high efficiency coding method which encodes speech at a coding speed of 16 kb/s, wherein due to use of a backward prediction method in regard to the LPC coefficients and the power of the input speech vector, transmission of the LPC coefficients codes and power codes of the input speech vector is unnecessary.
  • FIG. 17 is a block diagram showing a constructional example of a speech coding apparatus utilizing the conventional LD-CELP coding method. In this FIG. 17, components which correspond to those shown in FIG. 15, will retain the original identifying numeral, and their description will not herein be repeated.
  • a LPC analyzing portion 16 linear prediction analysis is not performed and the LPC coefficients of the synthesis filter 3 are not calculated for the input speech data, inputted from the input terminal 1, which is in the frame currently undergoing quantization. Instead, a high-order linear prediction analysis of the 50th order, including the pitch periodicity of the speech, is performed, and the LPC coefficients of the synthesis filter 3 are calculated and determined for the previously processed output vector of the synthesis filter 3. In this manner, the determined LPC coefficients are set into synthesis filter 3.
  • this speech coding apparatus after the calculation of the power of the input speech data in the frame undergoing quantization, in the power quantizing portion 10, the quantization of this power is not performed as in the speech coding apparatus shown in FIG. 15. Instead, in a gain adapting portion 17, linear prediction analysis is performed for the previously processed power of the output vector from the gain portion 8, and the power (in other words, the predicted gain) to be provided to the noise waveform vector selected in the current frame operation, is calculated, determined and then set into the predicted gain portion 18.
  • the predicted gain set by the gain adapting portion 17 is multiplied by the noise waveform vector which is selected and outputted from the random codebook 7 by the distortion power calculating portion 13. Subsequently, the gain set by the distortion power calculating portion 13 is multiplied by the output vector from the predicted gain portion 18 in the gain portion 8, and then outputted.
  • the output vector of the gain portion 8 is then supplied as an excitation vector to the synthesis filter 3, and a synthetic speech vector is synthesized in the synthesis filter 3 based on the set LPC coefficients.
  • the synthetic speech vector is subtracted from the input speech vector, and the distortion data are calculated.
  • the power of the distortion data outputted from the perceptual weighting filter 12 is calculated, the noise waveform vector, which will minimize the power of the distortion data, is selected from the random codebook 7, and the gain is then set in the gain portion 8.
  • the codes and gains selected according to the noise waveform vectors are converted into codes of bit series, outputted and then transmitted.
  • the decoded speech in the CELP speech coding, linear prediction analysis is performed, the LPC coefficients of the synthesis filter 3 are calculated and these LPC coefficients are then quantized only for the input speech data in the current frame undergoing quantization. Therefore, a drawback exists in that in order to obtain, at the transmission receiver, high quality speech which is decoded (hereafter referred to as "the decoded speech"), a large number of bits are necessary for the LPC coefficients quantization.
  • the power of the input speech vector is quantized, and the code selected in response to the quantized power of the input speech vector is transmitted as the coding signal, thus in the case where a transmission error of the code occurs in the transmission line, problems exist in that undesired speech is generated in the unvoiced intervals of the decoded speech, and the desired speech is frequently interrupted, thereby creating decoded speech of inferior quality.
  • quantization of the power of the input speech vector is performed using a limited number of bits, thus in the case where the magnitude of the input speech vector is small, a disadvantage exists in that the quantized noise increases.
  • the noise waveform vector is represented by one noise waveform vector stored in one random codebook 7, and the code selected in response to this noise waveform vector is transmitted as the coding signal, thus in the case where an transmission error of the code occurs in the transmission line, a completely different noise waveform vector is used in the speech decoding apparatus of the transmission receiver, thereby creating decoded speech of inferior quality.
  • the noise waveform vector to be stored in the random codebook uses a speech data base in which a large amount of actual speech data is stored, and performs learning so as to match this actual speech data.
  • the noise waveform vector is represented by one noise waveform vector of one random codebook 7
  • a large storage capacity is required, and thus the size of the codebook becomes significantly large. Consequently, disadvantages exist in that the aforementioned learning is not performed, and the noise waveform vector is not matched well with the actual speech data.
  • the pitch period vector and the noise waveform vector which will minimize the power of the distortion data are selected from the adaptive codebook 5 and the random codebook 7 respectively.
  • the power of the distortion data d shown in a formula (1) below, in a closed loop formed by means of structural elements 3, 5 ⁇ 9, and 11 ⁇ 13, or structural elements 3, 5, 7, 9, 11 ⁇ 13, and 15, must be calculated in the distortion power calculating portion 13 for all pitch period vectors and noise waveform vectors stored in the adaptive codebook 5 and the random codebook 7 respectively, there exist disadvantages in that enormous computational complexity is required.
  • the codebook 20 is formed from a plurality of codebooks, and in the coding portion in the LSP coefficient quantizing portion 4, the quantization error occurring in the vector quantization of a certain step is used as the input vector in the vector quantization of the next step.
  • the output vector is then formed by adding a plurality of the LSP codevectors selected from the plurality of the codebooks. In this manner, the vector quantization becomes possible while restricting the storage capacity and computational complexity to realistic ranges.
  • this multistage vector quantization method a distortion of significant proportion is observed when compared with the ideal onestage vector quantization method.
  • the LSP parameters must exist within the stable triangular region A1 shown in FIG. 19 according to the formula (2).
  • the expectation of the LSP parameters existing in the inclined region labeled A2 is high.
  • the LSP coding vector is represented as the sum of two vectors.
  • the codebook 20 is thus formed from a first codebook #1 and a second codebook #2.
  • step SA1 a 3-bit first codebook #1 similar to the input vector is formed.
  • step SA2 second vector quantization of the quantization error which occurred during quantization in step SA1 is performed. Namely, in step SA2 shown in FIG.
  • step SA3 the group of the reconstructed vectors V2 existing within the circular region shown in FIG. 22 (i.e. the contents of the second codebook #2) is centrally combined with the reconstructed vector V1, selected through the first vector quantization, thereby forming an output point.
  • step SA3 when two output vectors of codebook #1 and codebook #2 respectively are added, an output point may be formed in a region which did not originally exist. Consequently, in step SA3, a judgment whether the added vector is stable or unstable is made, with unstable vectors being excluded from the process.
  • step SA4 the distortion of the input vector and the aforementioned reconstructed vector is calculated. Subsequently, in step SA5, a vector is determined which will minimize the aforementioned distortion, and its code is transmitted to the decoding portion in the LSP coefficients quantizing portion 4.
  • step SA6 the codebook #1 is used to determine a first output vector, and in step SA7, a second output vector contained in the codebook #2, is added to this aforementioned first output vector, thereby yielding the final output vector.
  • the present invention provides a speech coding method for coding speech data comprising a plurality of samples as a unit of a frame operation wherein: the plurality of samples of speech data are analyzed by a linear prediction analysis and thereby prediction coefficients are calculated, and quantized; the quantized prediction coefficients are set in a synthesis filter; the synthesized speech vector is synthesized by exciting the synthesis filter with a pitch period vector which is selected from an adaptive codebook in which a plurality of pitch period vectors are stored, and which is multiplied by a first gain, and with a noise waveform vector which is selected from a random codebook in which a plurality of the noise waveform vectors are stored, and which is multiplied by a second gain; and wherein said method comprises choosing said first and second gain at the same time; providing a multiplier of multiplying the selected noise waveform vector by a predicted gain; and predicting said predicted gain which is to be multiplied by the noise waveform vector selected in a subsequent frame operation, and is based
  • the present invention provides a speech coding apparatus for coding speech data comprising a plurality of samples as a unit of a frame operation wherein: the plurality of samples of speech data are analyzed by a linear prediction analysis and thereby prediction coefficients are calculated and quantized; the quantized prediction coefficients are set in a synthesis filter; the synthetic speech vector is synthesized by exciting the synthesis filter with a pitch period vector which is selected from an adaptive codebook in which a plurality of pitch period vectors are stored, and which is multiplied by a first gain, and with a noise waveform vector which is selected from a random codebook in which a plurality of the noise waveform vectors are stored, and which is multiplied by a second gain; and wherein said apparatus comprises a gain predicting portion for multiplying said selected noise waveform vector by a predicted gain; a gain portion for multiplying said selected pitch period vector and an output vector derived from said gain predicting portion using said first and second gain, respectively, a distortion calculator for respectively selecting said pitch period vector and said noise wave
  • the present invention provides a parameter coding method of speech for quantizing parameters such as spectral envelope information and power information at a unit of a frame operation comprising a plurality of samples of speech data, wherein said method comprises the steps of, in a coding portion, (a) wherein said parameter is quantized, representing the resultant quantized parameter vector by the weighted mean of a prospective parameter vector selected from a parameter codebook in which a plurality of the prospective parameter vectors are stored in the current frame operation and a part of the prospective parameter vector selected from said parameter codebook in the previous frame operation, (b) selecting said prospective parameter vector from said parameter codebook so that a quantization distortion between said quantized parameter vector and an input parameter vector, is minimized, and (c) transmitting a vector code corresponding to the selected prospective parameter vector; and in a decoding portion, (a) calculating the weighted mean of the prospective parameter vector selected from said parameter codebook in the current frame operation corresponding to the transmitted vector code and the prospective parameter vector in the previous frame operation, and
  • the present invention provides a parameter coding apparatus of speech for quantizing parameters such as spectral envelope information and power information as a unit of a frame operation comprising a plurality of samples of speech data
  • said apparatus comprises a coding portion comprising, (a) a parameter codebook for storing a plurality of prediction parameter vectors, and (b) a vector quantization portion for calculating the weighted mean of the prospective parameter vector selected from said parameter codebook in the current frame operation, the part of the prospective parameter vector selected from said parameter codebook in the previous frame operation, using the resultant vector as the resultant quantized parameter vector of the quantization of prediction coefficients, selecting said prospective parameter vector from said parameter codebook so that a quantization distortion between said quantized parameter vector and an input parameter vector is minimized, and transmitting a vector code corresponding to the selected prospective parameter vector; and a decoding portion for calculating the weighted mean of the prospective parameter vector selected from said parameter codebook in the current frame operation corresponding to the transmitted vector code and the prospective parameter vector in the previous frame operation
  • the coding portion represents the resultant quantized parameter vector by the weighted mean of the prospective parameter vector selected from the parameter codebook in the current frame operation and the part of the prospective parameter vector selected from the parameter codebook in the previous frame operation. Then the coding portion selects the prospective parameter vector from the parameter codebook so that the quantization distortion between the quantized parameter vector and the input parameter vector is minimized. Furthermore, the coding portion transmits the vector code corresponding to the selected prospective parameter vector. Moreover the decoding portion calculates the weighted mean of the prospective parameter vector selected from the parameter codebook in the current frame operation corresponding to the transmitted vector code, and the prospective parameter vector in the previous frame operation, and outputs the resultant vector.
  • the present invention since only the code corresponding to one parameter codebook is transmitted to each frame, even if the frame length is shortened, the amount of transmitted information remains small. Additionally, the quantization distortion may be reduced when the continuity with the previous frame is high. As well, even in the case where the coding errors occur, since the prospective parameter vector in the current frame operation is equalized with one in the previous frame operation, the effect of the coding errors is small. Moreover, the effect of coding errors in the current frame operation can only extend up to two frames operation fore. If coding errors can be detected using a redundant code, the parameter with errors is excluded, and by calculating the mean described above, the effect of errors can also be reduced.
  • the present invention provides a multistage vector quantizing method for selecting the prospective parameter vector from a parameter codebook so that the quantization distortion between the prospective parameter vector and an input parameter vector becomes minimized, a vector code corresponding to the selected prospective parameter vector is transmitted, and wherein said method comprises the steps of, in a coding portion, (a) representing said prospective parameter vector by the sum of subparameter vectors respectively selected from stages of the subparameter codebooks, (b) respectively selecting subparameter vectors from stages of said subparameter codebooks, (c) adding subparameter vectors selected to obtain the prospective parameter vector in the current frame operation, (d) judging whether or not said prospective parameter vector in the current frame operation is stable, (e) converting said prospective parameter vector into a new prospective parameter vector so that said prospective parameter vector in the current frame operation becomes stable using the fixed rule in the case where said prospective parameter vector in the current frame operation is not stable, (f) selecting the prospective parameter vector from said parameter codebook so that said quantization distortion is minimized, and (g) transmitting a vector
  • the present invention provides a multistage vector quantizing apparatus for selecting the prospective parameter vector from a parameter codebook so that the quantization distortion between the prospective parameter vector and an input parameter vector becomes minimized, and transmitting a vector code corresponding to the selected prospective parameter vector
  • said apparatus comprises said parameter codebook comprising stages of subparameter codebooks in which subparameter vectors are respectively stored, a coding portion comprising a vector quantization portion for respectively selecting subparameter vectors from stages of said subparameter codebooks, and adding the selected subparameter vectors to obtain the prospective parameter vector in the current frame operation, judging whether or not said prospective parameter vector in the current frame operation is stable, converting said prospective parameter vector into a new prospective parameter vector so that said prospective parameter vector in the current frame operation becomes stable using the fixed rule in the case where said prospective parameter vector in the current frame operation is not stable, selecting the prospective parameter vector from said parameter codebook so that said quantization distortion is minimized, and transmitting a vector code corresponding to the selected prospective parameter vector; and a decoding portion for respectively selecting subparameter
  • the output point is examined to determine whether or not it is the probable output point (determining whether it is stable or unstable).
  • this vector is converted into a new output vector in the region which always exist using the fixed rule, and then quantized. In this manner, unselected combinations of codes are eliminated, and the quantization distortion may be reduced.
  • unstable, useless output vectors occurring after the first stage of the multistage vector quantization are converted using the fixed rule, into effective output vectors which may then be used.
  • advantages such as a greater reduction of the quantization distortion from an equivalent amount of information, as compared with the conventional methods may be obtained.
  • FIG. 1 (A) is a block diagram showing a part of a construction of a speech coding apparatus according to a preferred embodiment of the present invention.
  • FIG. 1 (B) is a block diagram showing a part of a construction of a speech coding apparatus according to a preferred embodiment of the present invention.
  • FIG. 2 is a block diagram showing a first construction of a vector quantization portion applied to a parameter coding method according to a preferred embodiment of the present invention.
  • FIG. 3(A) is a block diagram showing a second construction of a vector quantization portion applied to a parameter coding method according to a preferred embodiment of the present invention.
  • FIG. 3(B) is a reference diagram for use in explaining an example of the operation of the vector quantization portion shown in FIG. 3(A).
  • FIG. 4(A) is a block diagram showing a third construction of a vector quantization portion applied to a parameter coding method according to a preferred embodiment of the present invention.
  • FIG. 4(B) is a reference diagram for use in explaining an example of the operation of the vector quantization portion shown in FIG. 4(A).
  • FIG. 5 is a block diagram showing a fourth construction of a vector quantization portion applied to a parameter coding method according to a preferred embodiment of the present invention.
  • FIG. 6 is a block diagram showing a fifth construction of a vector quantization portion applied to a parameter coding method according to a preferred embodiment of the present invention.
  • FIG. 7 shows an example of a construction of the LSP codebook 37.
  • FIG. 8 is a flow chart for use in explaining a multistage vector quantization method according to a preferred embodiment of the present invention.
  • FIG. 9 shows the conversion of a reconstructed vector according to the preferred embodiment shown in FIG. 8.
  • FIG. 10 is a block diagram showing a sixth construction of a vector quantization portion applied to a parameter coding method according to a preferred embodiment of the present invention.
  • FIG. 11 shows an example of a construction of a vector quantization gain searching portion 65.
  • FIG. 12 shows an example of the SN characteristics plotted against the transmission line error percentage in a speech coding apparatus according to the conventional art, and one according to a preferred embodiment of the present invention.
  • FIG. 13 shows an example of a construction of a vector quantization codebook 31.
  • FIG. 14 shows an example of opinion values of decoded speech plotted against various evaluation conditions in a speech coding apparatus according to a preferred embodiment of the present invention.
  • FIG. 15 is a block diagram showing a constructional example of a speech coding apparatus utilizing a conventional CELP coding method.
  • FIG. 16 is a block diagram showing a constructional example of a speech coding apparatus utilizing the a conventional VSELP coding method.
  • FIG. 17 is a block diagram showing a constructional example of a speech coding apparatus utilizing a conventional LD-CELP coding method.
  • FIG. 18 is a block diagram showing a constructional example of a conventional vector quantization portion.
  • FIG. 19 shows the existence region of a two-dimensional LSP parameter according to a conventional multistage vector quantization method.
  • FIG. 20 is a flow chart for use in explaining a conventional multistage vector quantization method.
  • FIG. 21 shows a reconstructed vector of a first stage, in the case where vector quantization of the LSP parameters shown in FIG. 19 is performed.
  • FIG. 22 shows a vector to which a reconstructed vector of a second stage has been added, in the case where vector quantization of the LSP parameters shown in FIG. 19 is performed.
  • FIGS. 23-27 are flow charts for use in explaining multistage vector quantization methods according to alternative embodiments of the present invention.
  • FIG. 28 is a flow chart for use in explaining a vector quantization gain searching method according to a preferred embodiment of the present invention.
  • FIGS. 1 (A) and (B) are block diagrams showing a construction of a speech coding apparatus according to a preferred embodiment of the present invention. An outline of a speech coding method will now be explained with reference to FIGS. 1(A) and 1(B).
  • the input speech data formed by sampling the analog speech signal at a sampling frequency of 8 kHz is inputted from an input terminal 21. Eighty samples are then obtained as one frame in one vector and stored in a buffer 22 as an input speech vector.
  • the frame is then further divided into two subframes, each comprising a unit of forty samples. All processes following this will be conducted in frame units or subframe units.
  • a soft limiting portion 23 the magnitude of the input speech vector outputted from the buffer 22 is checked using a frame unit, and in the case where the absolute value of the magnitude of the input speech vector is greater than a previously set threshold value, compression is performed. Subsequently, in an LPC analyzing portion 24, linear prediction analysis is performed and the LPC coefficients are calculated for the input speech data of the plurality of samples outputted from the soft limiting portion 23. Following this, in an LSP coefficient quantizing portion 25, the LPC coefficients are quantized, and then set into a synthesis filter 26.
  • a pitch period vector and a noise waveform vector selected by a distortion power calculating portion 35 are outputted from an adaptive codebook searching portion 27 and a random codebook searching portion 28, respectively, and the noise waveform vector is then multiplied by the predicted gain set by to a gain adapting portion 29 in a predicted gain portion 30.
  • linear prediction analysis is performed based on the power of the output vector from a vector quantization gain codebook 31 in the current frame operation, and the stored power of the output vector of the random codebook component of the vector quantization gain codebook 31 which was used in the previous frame operation.
  • the power (namely the predicted gain) to be multiplied by the noise waveform vector selected in the subsequent frame operation is then calculated, determined and set into the predicted gain portion 30.
  • the selected pitch period vector and the output vector of the predicted gain portion 30 is determined in the distortion power calculating portion 35, multiplied, in subgain codebooks 31a and 31b of the vector quantization gain codebook 31, by the gains selected from these subgain codebooks 31a and 31b, and then outputted.
  • the output vectors of the subgain codebooks 31a and 31b are summed in an adder 32, and the resultant output vector of the adder 32 is supplied as an excitation vector to the synthesis filter 26.
  • the synthetic speech vector is then synthesized in the synthesis filter 26.
  • a subtracter 33 the synthetic speech vector is subtracted from the input speech vector, and the distortion data is calculated.
  • this distortion data is weighted in a perceptual weighting filter 34 according to the coefficients corresponding to human perceptual characteristics, the power of the distortion data outputted from the perceptual weighting filter 34 is calculated in the distortion power calculating portion 35.
  • the pitch period vector and noise waveform vector which will minimize the aforementioned power of the distortion data, are selected respectively from the adaptive codebook searching portion 27 and the noise codebook searching portion 28, and the gains of the subgain codebooks 31a and 31b are then designated.
  • a code outputting portion 36 the respective codes and gains selected according to the LPC coefficients, the pitch period vector and the noise waveform vector are then converted into codes of bit series, and when necessary, error correction codes are added and then transmitted.
  • the local decoding portion LDEC in order to prepare for the process of the subsequent frame in the coding apparatus of the present invention, uses the same data as that outputted and transmitted from each structural component shown in FIG. 1 to the decoding apparatus, and synthesizes a speech decoding vector.
  • the LPC coefficient quantizing portion 25 the LPC coefficients obtained in the LPC analyzing portion 24 are first converted to LSP parameters, quantized, and these quantized LSP parameters are then converted back into the LPC coefficients.
  • the LPC coefficients obtained by means of this series of processes, are thus quantized; LPC coefficients may be converted into LSP parameters using, for example, the Newton-Raphson method. Since a short frame length of 10 ms and a high correlation between each frame, by utilizing these nature, a quantization of the LSP parameters is performed using a vector quantization method.
  • the LSP parameters are represented by a weighted mean vector calculated from a plurality of vectors of past and current frames.
  • the output vectors in the past frame operation are used without variation; however, in the present invention, among the vectors formed through calculation of the weighted mean, only vectors updated in the immediately preceding frame operation are used. Furthermore, in the present invention, among the vectors formed through calculation of the weighted mean, only vectors unaffected by coding errors and vectors in which coding errors have been detected and converted are used.
  • the present invention is also characterized in that the ratio of the weighted mean is either selected or controlled.
  • FIG. 2 shows a first construction of a vector quantizing portion provided in the LPC coefficients quantizing portion 25.
  • An LSP codevector V k-1 (k is the frame number), produced from a LSP codebook 37 in the frame operation immediately preceding the current frame operation, is multiplied in a multiplier 38 by a multiplication coefficient (1-g), and then supplied to one input terminal of an adder 39.
  • a mark g represents a constant which is determined by the ratio of the weighted mean.
  • An LSP codevector V k produced from the LSP codebook 37 in the current frame operation is supplied to each input terminal of a transfer switch 40.
  • This transfer switch 40 is activated in response to the distortion calculation result by a distortion calculating portion 41.
  • the selected LSP codevector V k is first multiplied by the multiplication coefficient g in a multiplier 42, and then supplied to the other input terminal of the adder 39. In this manner, the output vectors of the multipliers 38 and 42 are summed in the adder 39, and the quantized LSP parameter vector ⁇ k of the frame number k is then outputted.
  • this LSP parameter vector ⁇ k may be expressed by the following formula (3).
  • the distortion calculating portion 41 the distortion data between an LSP parameter vector ⁇ k of the frame number k before quantization and the LSP parameter vector ⁇ k of the frame number k following quantization, is calculated, and the transfer switch 40 is activated such that this distortion data is minimized.
  • the code for the LSP codevector V k selected by the distortion calculator 41 is outputted as a code S 1 .
  • the LSP codevector V k produced from the LSP codebook 37 in the current frame operation is employed in the subsequent frame operation as an LSP codevector V k-1 , which is produced from the LSP codebook 37 in the previous frame operation
  • FIG. 23 shows a flowchart where steps SC1-SC7 portray the operation of the vector quantizing portion described above and shown in FIG. 2.
  • three types of codebooks 37, 43, and 44 are used corresponding to the frame number.
  • the quantized LSP parameter vector ⁇ k may be calculated using a mean of three vectors in the frames in formula (4) below. ##EQU1##
  • An LSP codevector V k-2 represents the LSP codevector produced from the LSP codebook 43 in the two frame operations prior to the current frame operation, while an LSP codevector V k-1 represents the LSP codevector produced from the LSP codebook 44 in the frame operation immediately preceding the current frame operation.
  • an LSP codevector which will minimize the distortion data between the LSP parameter vector ⁇ k of the frame number k before quantization and the LSP parameter vector ⁇ k of the frame number k (the kth frame) following quantization, is selected from the LSP codebook 37.
  • the code corresponding to the selected LSP codevector V k is then outputted as the code S1.
  • the LSP codevector V k-1 may also be used in the subsequent frame operation, and similarly the LSP codevector V k may be used in the next two frame operations.
  • the LSP codevector V k may be determined at the kth frame operation, if this decision may be delayed, the quantization distortion can be reduced when this decision is delayed in consideration of the LSP parameter vectors ⁇ k+1 and ⁇ k+2 , appearing in the subsequent frame and two frame operations later.
  • FIG. 24 shows a flowchart where steps SD1-SD6 portray the operation of the LSP parameter vector quantization method described with reference to FIGS. 3A and 3B.
  • This vector quantization method is similar to the vector quantization method shown in FIGS. 3(A) and 3(B), however, the quantized LSP parameter vector ⁇ k of the frame number k is expressed using the following formula (5). ##EQU2##
  • FIG. 25 shows a flowchart where steps SE1-SE8 portray the operation of the LSP parameter vector quantization method described with reference to FIG. 4.
  • the codebooks 37, 43, and 44 are presented separately; however, it is also possible for these codebooks to be combined into one common codebook as well.
  • the ideal LSP parameter vector ⁇ k is previously provided, and a method is employed which determines the LSP parameter vector ⁇ k quantized using the mean calculated in the parameter dimensions.
  • the LSP parameters there exists a method for determining the LSP parameters of the current frame by analyzing a plurality of times the distortion data outputted from an inverse filter, in which the LSP parameters determined in a previous frame operation is set.
  • the mean calculated from the coefficients of the polynomial expressions of the individual synthesis filters becomes the final synthesis filter coefficients.
  • the product of the terms of the individual polynomial expressions becomes the final synthesis filter polynomial expression.
  • the LSP codevector is selected so that the distortion data between an expected value ⁇ * k in the local decoding portion LDEC in consideration for a coding error rate, instead of the output vector, the LSP parameter vector ⁇ k in FIG. 2, and the input vector, the LSP parameter vector ⁇ k are minimized.
  • This expected value ⁇ * k may be estimated using formula (6) below.
  • ⁇ represents the coding error rate in the transmission line (a 1 bit error rate), and m represents the transmission bit number per a vector).
  • ⁇ e erepresents m types of vectors which are outputted in the case where an error occurs in only one bit of m pieces of the transmission line codes corresponding to the LSP parameter vector ⁇ k and a second term of the righthand side of the equation represents the sum of these m types of vectors ⁇ e .
  • FIG. 5 a second construction of a vector quantization portion provided in the LPC coefficients quantizing portion 25 is shown.
  • components which correspond to those shown in FIG. 2 will retain the original identifying numeral, and their description will not herein be repeated.
  • a constant g determined from the ratio of the weighted mean is not fixed, rather a ratio constant g k is designated according to each LSP code V k stored in the LSP codebook 37.
  • each LSP codevector V k outputted from the LSP codebook 37 is multiplied by the appropriate multiplication coefficient g 1 , g 2 , . . .
  • FIG. 26 shows a flowchart where steps SF1-SF7 portray the operation of the vector quantization portion described above and shown in FIG. 5.
  • the distortion calculating portion 41 is constructed in a manner such that the LSP codevector V k , which will minimize the distortion data between the quantized LSP parameter vector ⁇ k outputted from the adder 39 and the LSP parameter vector ⁇ k before quantization, are selected by transferring the transfer switch 46, and the corresponding multiplication coefficient g k are selected.
  • the aforementioned construction is designed such that the ratio (1-g k ) supplied to the multiplier 47 is interlocked and changed by means of the transfer switch 46.
  • the quantized LSP parameter vector ⁇ k may be expressed using the following formula (7).
  • the multiplication coefficient g k is a scalar value corresponding to the LSP codevector V k ; however, it is also possible to assemble a plurality of the LSP codevectors as one group, and have this scalar value correspond to each of these types of groups. In addition, it is also possible to proceed in the opposite manner by setting the multiplication coefficient at each component of the LSP codevector.
  • the LSP codevector V k-1 produced from the LSP codebook 37 in the previous frame operation is given, and in order to minimize the distortion data between the quantized LSP parameter vector ⁇ k and the LSP parameter vector ⁇ k before quantization, the most suitable combination of the ratio g k which is the ratio of the weighted mean between the LSP codevector V k produced from the LSP codebook 37 in the current frame operation and the LSP codevector V k-1 produced from the LSP codebook 44 in the previous frame operation, and the LSP codevector V k , is selected.
  • FIG. 6 shows a third construction of a vector quantization portion provided in the LSP coefficient quantizing portion 25.
  • the vector quantization portion shown in FIG. 6 is characterized in that the ratio value of a plurality of different types of weighted means is set independently from the LSP codevectors.
  • the LSP codevector V k-1 produced from the LSP codebook 37 in the frame operation immediately prior to the current frame operation is multiplied, in multipliers 47 and 48, by the multiplication coefficients (1-g 1 ) and (1-g 2 ) respectively, and then supplied to the input terminals T a and T b of a transfer switch 49.
  • the transfer switch 49 is activated in response to the distortion calculation resulting by the distortion calculating portion 41, and the output vector from either multiplier 47 or 48 is selected, and supplied to one input terminal of the adder 39 via a common terminal T c .
  • an LSP codevector V k produced from the LSP codebook 37 in the current frame operation, is supplied to each input terminal of the transfer switch 40.
  • the transfer switch 40 is activated in the same manner as the transfer switch 49, in response to the distortion calculation result by the distortion calculator 41. In this manner, the selected LSP codevector V k is multiplied, in multipliers 50 and 51, by multiplication coefficients g 1 and g 2 respectively, and then supplied to input terminals T a and T b of a transfer switch 52.
  • the transfer switch 52 is activated in the same manner as the transfer switches 40 and 49, in response to the distortion calculation result by the distortion calculator 41, and the output vector from either multiplier 50 or 51 is selected, and supplied to one input terminal of the adder 39 via the common terminal T c .
  • this LSP parameter vector ⁇ k may be expressed by the following formula (8).
  • m is 1 or 2.
  • the distortion data between the LSP parameter vector ⁇ k of the frame number k before quantization and the LSP parameter vector ⁇ k of the frame number k after quantization are calculated in th e distortion calculating portion 41, and the transfer switches 49 and 52 are activated in a manner such that this distortion data is minimized.
  • the code S1 the code of the selected LSP codevector V k , and the selection information S2, indicating which the output vectors from each of the multipliers 47 and 48, and 50 and 51 will be used, are outputted from the distortion calculating portion 41.
  • the LSP codevector V k is expressed as the sum of two vectors.
  • the LSP codebook 37 is formed from a first stage LSP codebook 37a, in which 10 vectors E 1 have been stored, and a second stage LSP codebook 37b1, which comprises two separate LSP codebooks each storing five vectors, a second stage low order LSP codebook 37b1 and a second stage high order LSP codebook 37b2.
  • the LSP codevector V k may be expressed using the following formulae (9) and (10).
  • an E 1n is an output vector of the first stage LSP codebook 25a, and n is 1 through 128. In other words, 128 output vectors E 1 are stored in the first stage LSP codebook 25a.
  • an E L2f is an output vector of the second stage low order LSP codebook 37b1 and an E H2f is an output vector of the second stage high order LSP codebook 37b2.
  • the vector quantization method (not shown in the FIGS.) used in this vector quantization portion reduces the effects of coding errors in the case where these errors are detected in the decoding portion. Similar to the vector quantization portion shown in FIG. 2, this method calculates, in the coding portion, the LSP vector V k which will minimize the distortion data. However, in the case where coding errors are detected or highly probable in either LSP codevector V k-1 in the previous frame operation in the decoding portion, or LSP codevector V k in the current frame operation, only in the decoding portion, this method calculates an output vector by reducing the ratio of the weighted mean of the LSP vectors incorporating the errors.
  • the LSP parameter vector ⁇ k may be expressed by formula (12) in order to reduce the effects of the transmission line errors from the previous frame. ##EQU3##
  • step SB1 the distortion calculating portion 41 selects a plurality of the output vectors E 1n similar to the LSP parameter vector ⁇ k from the first stage LSP codebook 37a, by means of appropriately activating the transfer switch 40.
  • step SB2 the distortion calculating portion 41 respectively adds to each of the selected high and low order output vectors E 1n , the output vectors E L2f and E H2f selected respectively from the second stage low order LSP codebook 37b1 and the second stage high order LSP codebook 37b2 of the second stage codebook 37b, and produces the LSP codevector V k .
  • the system then proceeds to step SB3.
  • step SB3 the distortion calculating portion 41 judges whether or not the LSP codevector V k obtained in step SB2 is stable. This judgment is performed in order to stabilize and activate the synthesis filter 26 (see FIG. 1) in which the aforementioned LSP codevector V k is set.
  • the values of the LSP parameters ⁇ 1 through ⁇ p forming p number of the LSP codevectors V k must satisfy the relationship shown in the aforementioned formula (2).
  • the distortion calculating portion 41 converts the output vector P into a new output vector P1, which is symmetrical in relation to the broken line L1 shown in FIG. 9 in order to achieve a stable situation.
  • the LSP codevector V k which is either stable or has been converted so as to stabilize, is multiplied respectively, in the multipliers 50 and 51, by the multiplication coefficients g 1 and g 2 .
  • the output vector of either multiplier 50 or 51 is then supplied to the other input terminal of the adder 39 via the transfer switch 52.
  • the LSP codevector V k-1 produced from the LSP codebook 37 in the frame operation immediately prior to the current frame operation, is multiplied, in the multipliers 47 and 48, by the multiplication coefficients (1-g 1 ) and (1-g 2 ) respectively, and the output vector of either multiplier 47 or 48 is then supplied to one input terminal of the adder 39 via the transfer switch 49.
  • the weighted mean of the output vectors of the transfer switches 49 and 52 are calculated, and the LSP parameter vector ⁇ k is outputted.
  • step SB4 the distortion calculator 41 calculates the distortion data between the LSP parameter vector ⁇ k and the LSP parameter vector ⁇ k , and the process moves to step SB5.
  • step SB5 the distortion calculating portion 41 judge whether or not the distortion data calculated in step SB4 is at a minimum. In the case where this judgment is "NO”, the distortion calculating portion 41 activates either transfer switch 49 or 51, returning the process to step SB2.
  • the aforementioned steps SB2 to SB5 are then repeated in regard to the plurality of output vectors E 1n selected in step SB1.
  • the distortion calculating portion 41 determines the LSP codevector V k , outputs this code as the code S 1 , outputs the selection information S 2 , and transmits them respectively to the decoding portion in the vector quantization portion.
  • the decoding portion comprises the LSP codebook 37 and the transfer switches 40, 49 and 52 shown in FIG. 6.
  • step SB6 the decoding portion activates the transfer switch 40 based on the transmitted code S 1 , and selects the output vector E 1n from the first stage codebook 37a.
  • step SB7 the decoding portion activates the transfer switch 40 based on the transmitted selection information S 2 to respectively select the output vectors E L2f and E H2f from the second stage low order LSP codebook 37b1 and the second stage high order LSP codebook 37b2 of the second stage codebook 37b, adds them to respectively the high and low order of the selected output vectors E 1n , and thereby produces the LSP codevector V k .
  • the system then proceeds to step SB8.
  • step SB8 the decoding portion judges whether or not the LSP codevector V k obtained in step SB7 is stable.
  • the decoding portion judges that the LSP codevector V k is unstable, as in step SB3 above, it converts the output vector P into a new output vector P1, which is symmetrical in relation to the broken line L1 shown in FIG. 9 in order to achieve a stable situation.
  • the LSP codevector V k which is either stable or has been converted so as to stabilize, may be used in the subsequent frame operation as the LSP codevector V k-1 .
  • the multistage vector quantization method shown above in FIG. 6 is characterized in that when the output vectors E L2f and E H2f selected respectively from the second stage low order LSP codebook 37b1 and the second stage high order LSP codebook 37b2 of the second stage codebook 37b, are summed, in the case where an unstable output vector is present, the output position is shifted, and the output vector P is converted into the output vector P1, which is symmetrical in relation to the broken line L1 shown in FIG. 9.
  • the diagonal line represents the set of values at which the LSP parameters ⁇ 1 and ⁇ 2 are equal.
  • FIG. 10 shows a fourth construction of a vector quantization portion provided in the LSP coefficient quantizing portion 25.
  • Adders 53 to 55, multipliers 56 to 61 and transfer switches 62 to 64 comprise the same functions as the adder 39, the multiplier 47 and the transfer switch 49, respectively.
  • the vector quantization portion shown in FIG. 10 calculates the LSP parameter vector ⁇ k , expressed in formula (13), using the weighted means of a plurality of the past LSP codevectors V k-4 to V k-1 and the current LSP codevector V k .
  • g 4m to g m are the constants of the weighted means, and m is 1 or 2.
  • FIG. 10 shows a flowchart where steps SG1-SG10 portray the operation of the vector quantization portion described above and shown in FIG. 10.
  • FIG. 11 shows a detailed block diagram of the vector quantization gain searching portion 65.
  • the linear prediction analysis is carried out for the power of the output vector from the vector quantization gain codebook 31 at the present operation, and for the power of the output vector of random codebook component from the vector quantization gain codebook 31, which is used in the past operation and is stored in the vector quantization gain codebook 31.
  • the gain adapting portion 29 the predicted gain by which the noise waveform vector which will be selected at a next frame operation, will multiply, is calculated and decided, and the decided predicted gain is set in the gain adapting portion 30.
  • the vector quantization gain codebook 31 is divided into subgain codebooks 31a and 31b to increase the quantization efficiency by the vector quantization and to decrease the effect on the decoded speech in the case where the error of the gain code is occurred in a transmission line.
  • the pitch period outputted from the adaptive codebook searching portion 27, is supplied to the subgain codebooks 31a and 31b in block of one-half, respectively, and the half of the output vector from the predicted gain portion 30 is supplied to the subgain codebooks 31a and 31b in block of one-half, respectively.
  • the gain multiplied by each of the vectors is selected as a block by the distortion power calculating portion 35 shown in FIG. 1 so that the distortion data that is the difference between an input speech vector and a synthesized speech vector, is minimized as a whole.
  • FIG. 28 shows a flowchart where steps SC7, SC5, SC6, and SD2 portray the operation of the vector quantization gain searching portion described above and shown in FIG. 11. Accordingly, it is possible to decrease the effect of the error in the transmission line.
  • FIG. 12 shows an example of signal-to-noise ratio (SNR) characteristics for the transmission error rate in the case of representing the gain by which the pitch period vector and the noise waveform vector is multiplied, respectively, by the output vector from the conventional gain codebook, and the case of representing one by the sum of the output vectors from two subgain codebooks.
  • SNR signal-to-noise ratio
  • a curve a shows the SNR characteristics according to the conventional gain codebook
  • a curve b shows one according to the subgain codebooks of this embodiment of the present invention.
  • the vector quantization gain codebook 31 is composed of the subgain codebooks 31a and 31b serially connected as shown in FIG. 13.
  • the gain by which the pitch period vector is multiplied is selected from ⁇ g p0 , g p1 , . . . , g pM ⁇ .
  • the gain by which the output vector of the predicted gain portion 30 is multiplied is selected from ⁇ g c0 , g c1 , . . . , g cM ⁇ .
  • the gain code of the pitch period vector is not at all affected by the transmission error of the gain code of the output vector from the predicted gain portion 30.
  • the transmission error of the gain code of the output vector from the predicted gain portion 30 also occurs.
  • the gain codes of these gains it is possible to decrease the effect of the transmission error of the gain code in the transmission line.
  • the pitch period vector and the noise waveform vector are respectively selected from among a plurality of the pitch period vectors and a plurality of the noise waveform vectors respectively stored in the adaptive codebook 27 and the random codebook 28 so that the power of the distortion d' represented by the formula (14), is minimized.
  • X T represents a target input speech vector used when the optimum vector is searched in the adaptive codebook searching portion 27 and the random codebook searching portion 28.
  • the target input speech vector X T is obtained by subtracting a zero input response vector X Z of the decoded speech vector which is decoded in the previous frame operation and is perceptually weighted in the perceptual weighting filter 34, from the input speech vector X W perceptually weighted in the perceptual weighting filter 34 as shown in formula (15).
  • the zero input response vector X Z is the component of the decoded speech vector operated until one frame before the current frame that affects the current frame, and is obtained by inputting a vector comprising a zero sequence into the synthesis filter 26.
  • the vector V' i is selected from each of the codebooks based on this correlation value X T T HV' i .
  • the distortion d' is not calculated for the entire vector V' i stored in each of codebooks, but only the correlation value is calculated for the entire vector V' i and the distortion d' is calculated for only the vector V' i having the large correlation value X T T HV' i .
  • the correlation calculation between the target input speech vector X T and the synthesis speech vector HV' is carried out.
  • the N times of the filtering calculation and the N times of preforming the correlation calculation are necessary for the calculation of the synthesis speech vector HV' because the number of the vector V' i is equal to the codebook size N.
  • a backward filtering disclosed in "Fast CELP Coding based on algebraic codes", Proc. ICASSP'87, pp. 1957-1960, J. P. Adoul, et al., is used.
  • X T T H is initially calculated and (X T T H)V' is calculated.
  • the correlation value X T T HV' i is obtained by filtering one time and performing the correlation calculation N times.
  • the arbitrary numbers of the vector V' i having the large correlation value X T T HV' i are selected and the filtering of the synthesis speech vector HV' i may be calculated only for the selected arbitrary number of the vector V' i . Consequently, it is possible to greatly decrease the computational complexity.
  • the adaptive codebook searching portion 27 comprises the adaptive codebook 66 and the pre-selecting. portion 68.
  • the adaptive codebook searching portion 37 the past waveform vector (pitch period vector) which is most suitable for the waveform of the current frame, is searched as a unit of a subframe.
  • Each of the pitch period vectors stored in the adaptive codebook 66 is obtained by passing the decoded speech vector through a reverse filter.
  • the coefficient of the reverse filter is the quantized coefficient
  • the output vector from the reverse filter is the residual waveform vector of the decoded speech vector.
  • the pre-selecting portion 68 the pre-selection of a prospect of the pitch period vector (hereafter referred to as a pitch prospect) to be selected is carried out twice.
  • M pieces for example, 16 pieces
  • the optimum pitch prospect among the pitch prospects selected in the pre-selecting portion 68 is decided as the pitch period vector to be outputted.
  • the optimum gain g' is set as shown a formula (17)
  • the above-mentioned formula (16) can be modified as shown a formula (18). ##EQU4##
  • the pitch prospect that the smallest distortion d' can be obtained is searched is equal to what the pitch prospect that the second term of the formula (18) is maximized is searched. Accordingly, the second term of the formula (18) is respectively calculated for the M pieces of the pitch prospect selected in the pre-selecting portion 68, and the pitch prospect which the calculating result is maximized, is decided as the pitch period vector HP to be outputted.
  • the random codebook searching portion 28 comprises a random codebook 67, and pre-selecting portions 69 and 70.
  • a waveform vector (a noise waveform vector) which is most suitable for the waveform of the current frame, is searched for among a plurality of the noise waveform vectors stored in the random codebook 67 as a unit of a subframe.
  • the random codebook 67 comprises subcodebooks 67a and 67b. In the subcodebooks 67a and 67b, a plurality of excitation vectors are stored, respectively.
  • the noise waveform vector C d is represented by the sum of two excitation vectors as shown in formula (19).
  • the excitation vectors C sub1p and C sub2q is represented by 7 bits, and the signs ⁇ 1 and ⁇ 2 is represented by 1 bit. If the noise waveform vector C d is represented by a single vector as in the conventional art, the excitation vectors C sub1p and C sub2q will be represented by 15 bits, and the signs ⁇ 1 and ⁇ 2 will be represented by 1 bit. Accordingly, because a large amount of memory is required for the random codebook, the codebook size is too large. However, as this embodiment, since the noise waveform vector C d is represented by the sum of the two excitation vectors C sub1p and C sub2q , the codebook size of the random codebook 67 can be greatly decreased compared with that of the conventional art.
  • the excitation vectors C sub1p and C sub2q are respectively pre-selected from the subcodebooks 67a and 67b.
  • the correlation value between the excitation vectors C sub1p and C sub2q and the target input speech vector X T are respectively calculated and the pre-selection of a prospect of the noise waveform vector C d (hereafter referred to as a random prospect) to be selected, is carried out.
  • the noise waveform vector is searched for by orthogonalizing each of the random prospects against the searched pitch period vector HP to increase quantization efficiency.
  • the orthogonalized noise waveform vector HC d ! against the pitch period vector HP is represented by formula (20). ##EQU5##
  • the pre-selection of the random prospect is carried out using the correlation value X T T HC d !.
  • the numerator term (HC d ) T HP of the second term is equivalent to (HP) T HC d .
  • the above-mentioned backward filtering is applied to the first term X T T HC d of the formula (21) and (HP) T HC d .
  • the noise waveform vector C d is the sum of the excitation vectors C sub1p and C sub2q
  • the correlation value X T T HC d ! is represented by formula (22).
  • the calculation shown by the formula (22) is carried out respectively for the excitation vectors C sub1p and C sub2q and the M pieces of the calculated correlation values whose value is large among these are respectively selected.
  • the random prospects comprising the most suitable combination are respectively chosen as a noise waveform vector to be outputted among each of the M pieces of the excitation vectors C sub1p and C sub2q selected in the pre-selecting portion 69 and 70.
  • the combination of the excitation vectors C sub1p and C sub2q which the second term of the formula (23) representing the distortion d" calculated using the target input speech vector X T and the random prospect is searched for. ##EQU7##
  • the calculation shown by the formula (23) may be carried out M 2 times on the whole.
  • the M pieces of the excitation vectors C sub1p and C sub2q are respectively pre-selected in the pre-selecting portions 69 and 70 and the optimum combination is selected among the M pieces of the pre-selected excitation vectors C sub1p and C sub2q , it is possible to further increase tolerance to the transmission error.
  • one noise waveform vector C d is represented by the two excitation vectors C sub1p and C sub2q , even if the error of either of the codes respectively corresponding to the excitation vectors C sub1p and C sub2q occurs in the transmission line, it is possible to compensate for the transmission error of one code with the other code.
  • the excitation vectors C sub1p and C sub2q having the high correlation with the target input speech vector are pre-selected by the pre-selection and then the optimum combination of the excitation vectors C sub1p and C sub2q is chosen as the noise waveform vector to be outputted, the noise waveform vector in which the transmission error has not occurred has a high correlation with the target input speech vector X T T . Consequently, in comparison with not carrying out the pre-selection, it is possible to decrease the effects of the transmission errors.
  • FIG. 14 shows a result in which the speech quality of the decoded speech was estimated by an opinion test in the case where the speech data are respectively coded and transmitted by the speech coding apparatus according to the conventional art and the present invention and are decoded by the speech decoding apparatus.
  • the speech quality of the decoded speech is depicted when the level of an input speech data in the speech coding apparatus is respectively set at 3 stages (A: large level, B: medium level, C: small level) in the case where transmission error has not occurred and the speech quality (see the mark D) of the decoded speech in the case where a random error ratio is 0.1%.
  • A large level
  • B medium level
  • C small level
  • oblique lined blocks show the result according to the conventional adaptive differential pulse coding modulation (ADPCM) method
  • crosshatched blocks show the result according to this embodiment of the present invention.
  • ADPCM adaptive differential pulse coding modulation

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Cited By (29)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5909663A (en) * 1996-09-18 1999-06-01 Sony Corporation Speech decoding method and apparatus for selecting random noise codevectors as excitation signals for an unvoiced speech frame
US5953698A (en) * 1996-07-22 1999-09-14 Nec Corporation Speech signal transmission with enhanced background noise sound quality
US5963896A (en) * 1996-08-26 1999-10-05 Nec Corporation Speech coder including an excitation quantizer for retrieving positions of amplitude pulses using spectral parameters and different gains for groups of the pulses
WO2000008633A1 (fr) * 1998-08-06 2000-02-17 Matsushita Electric Industrial Co., Ltd. Generateur de signaux d'excitation, codeur vocal et decodeur vocal
US6073092A (en) * 1997-06-26 2000-06-06 Telogy Networks, Inc. Method for speech coding based on a code excited linear prediction (CELP) model
US6138089A (en) * 1999-03-10 2000-10-24 Infolio, Inc. Apparatus system and method for speech compression and decompression
US6272196B1 (en) * 1996-02-15 2001-08-07 U.S. Philips Corporaion Encoder using an excitation sequence and a residual excitation sequence
US20020055836A1 (en) * 1997-01-27 2002-05-09 Toshiyuki Nomura Speech coder/decoder
US20020087863A1 (en) * 2000-12-30 2002-07-04 Jong-Won Seok Apparatus and method for watermark embedding and detection using linear prediction analysis
US20020103638A1 (en) * 1998-08-24 2002-08-01 Conexant System, Inc System for improved use of pitch enhancement with subcodebooks
US20030078774A1 (en) * 2001-08-16 2003-04-24 Broadcom Corporation Robust composite quantization with sub-quantizers and inverse sub-quantizers using illegal space
US20030083865A1 (en) * 2001-08-16 2003-05-01 Broadcom Corporation Robust quantization and inverse quantization using illegal space
US20030105624A1 (en) * 1998-06-19 2003-06-05 Oki Electric Industry Co., Ltd. Speech coding apparatus
US6594626B2 (en) * 1999-09-14 2003-07-15 Fujitsu Limited Voice encoding and voice decoding using an adaptive codebook and an algebraic codebook
US20040093207A1 (en) * 2002-11-08 2004-05-13 Ashley James P. Method and apparatus for coding an informational signal
US6738733B1 (en) * 1999-09-30 2004-05-18 Stmicroelectronics Asia Pacific Pte Ltd. G.723.1 audio encoder
US20070233472A1 (en) * 2006-04-04 2007-10-04 Sinder Daniel J Voice modifier for speech processing systems
US20080162150A1 (en) * 2006-12-28 2008-07-03 Vianix Delaware, Llc System and Method for a High Performance Audio Codec
US20080312917A1 (en) * 2000-04-24 2008-12-18 Qualcomm Incorporated Method and apparatus for predictively quantizing voiced speech
US20090024395A1 (en) * 2004-01-19 2009-01-22 Matsushita Electric Industrial Co., Ltd. Audio signal encoding method, audio signal decoding method, transmitter, receiver, and wireless microphone system
US20090198491A1 (en) * 2006-05-12 2009-08-06 Panasonic Corporation Lsp vector quantization apparatus, lsp vector inverse-quantization apparatus, and their methods
US20090299736A1 (en) * 2005-04-22 2009-12-03 Kyushu Institute Of Technology Pitch period equalizing apparatus and pitch period equalizing method, and speech coding apparatus, speech decoding apparatus, and speech coding method
US20100023324A1 (en) * 2008-07-10 2010-01-28 Voiceage Corporation Device and Method for Quanitizing and Inverse Quanitizing LPC Filters in a Super-Frame
US20100049508A1 (en) * 2006-12-14 2010-02-25 Panasonic Corporation Audio encoding device and audio encoding method
US20100076754A1 (en) * 2007-01-05 2010-03-25 France Telecom Low-delay transform coding using weighting windows
US20110004469A1 (en) * 2006-10-17 2011-01-06 Panasonic Corporation Vector quantization device, vector inverse quantization device, and method thereof
US20160302102A1 (en) * 2006-05-12 2016-10-13 Microsoft Technology Licensing, Llc Signaling to application lack of requested bandwidth
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US5751903A (en) * 1994-12-19 1998-05-12 Hughes Electronics Low rate multi-mode CELP codec that encodes line SPECTRAL frequencies utilizing an offset
US5648989A (en) * 1994-12-21 1997-07-15 Paradyne Corporation Linear prediction filter coefficient quantizer and filter set
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US5664055A (en) * 1995-06-07 1997-09-02 Lucent Technologies Inc. CS-ACELP speech compression system with adaptive pitch prediction filter gain based on a measure of periodicity
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US6131084A (en) * 1997-03-14 2000-10-10 Digital Voice Systems, Inc. Dual subframe quantization of spectral magnitudes
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JP3842432B2 (ja) * 1998-04-20 2006-11-08 株式会社東芝 ベクトル量子化方法
US6556966B1 (en) 1998-08-24 2003-04-29 Conexant Systems, Inc. Codebook structure for changeable pulse multimode speech coding
US6493665B1 (en) * 1998-08-24 2002-12-10 Conexant Systems, Inc. Speech classification and parameter weighting used in codebook search
US6173257B1 (en) 1998-08-24 2001-01-09 Conexant Systems, Inc Completed fixed codebook for speech encoder
US6714907B2 (en) 1998-08-24 2004-03-30 Mindspeed Technologies, Inc. Codebook structure and search for speech coding
SE519563C2 (sv) * 1998-09-16 2003-03-11 Ericsson Telefon Ab L M Förfarande och kodare för linjär prediktiv analys-genom- synteskodning
CA2259094A1 (en) * 1999-01-15 2000-07-15 Universite De Sherbrooke A method and device for designing and searching large stochastic codebooks in low bit rate speech encoders
US8195452B2 (en) 2008-06-12 2012-06-05 Nokia Corporation High-quality encoding at low-bit rates
EP2304722B1 (de) 2008-07-17 2018-03-14 Nokia Technologies Oy Verfahren und vorrichtung zur schnellen suche des nächsten nachbarn für vektorquantisierer
US9842598B2 (en) 2013-02-21 2017-12-12 Qualcomm Incorporated Systems and methods for mitigating potential frame instability
CN104751850B (zh) * 2013-12-25 2021-04-02 北京天籁传音数字技术有限公司 一种用于音频信号的矢量量化编解码方法及装置

Citations (12)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP0296763A1 (de) * 1987-06-26 1988-12-28 AT&T Corp. CELP Vocoder und Anwendungsverfahren
US4860355A (en) * 1986-10-21 1989-08-22 Cselt Centro Studi E Laboratori Telecomunicazioni S.P.A. Method of and device for speech signal coding and decoding by parameter extraction and vector quantization techniques
US4975956A (en) * 1989-07-26 1990-12-04 Itt Corporation Low-bit-rate speech coder using LPC data reduction processing
US4991214A (en) * 1987-08-28 1991-02-05 British Telecommunications Public Limited Company Speech coding using sparse vector codebook and cyclic shift techniques
US5010574A (en) * 1989-06-13 1991-04-23 At&T Bell Laboratories Vector quantizer search arrangement
US5230037A (en) * 1990-10-16 1993-07-20 International Business Machines Corporation Phonetic hidden markov model speech synthesizer
US5305332A (en) * 1990-05-28 1994-04-19 Nec Corporation Speech decoder for high quality reproduced speech through interpolation
US5321793A (en) * 1992-07-31 1994-06-14 SIP--Societa Italiana per l'Esercizio delle Telecommunicazioni P.A. Low-delay audio signal coder, using analysis-by-synthesis techniques
US5377301A (en) * 1986-03-28 1994-12-27 At&T Corp. Technique for modifying reference vector quantized speech feature signals
US5396576A (en) * 1991-05-22 1995-03-07 Nippon Telegraph And Telephone Corporation Speech coding and decoding methods using adaptive and random code books
US5426460A (en) * 1993-12-17 1995-06-20 At&T Corp. Virtual multimedia service for mass market connectivity
US5432883A (en) * 1992-04-24 1995-07-11 Olympus Optical Co., Ltd. Voice coding apparatus with synthesized speech LPC code book

Patent Citations (12)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US5377301A (en) * 1986-03-28 1994-12-27 At&T Corp. Technique for modifying reference vector quantized speech feature signals
US4860355A (en) * 1986-10-21 1989-08-22 Cselt Centro Studi E Laboratori Telecomunicazioni S.P.A. Method of and device for speech signal coding and decoding by parameter extraction and vector quantization techniques
EP0296763A1 (de) * 1987-06-26 1988-12-28 AT&T Corp. CELP Vocoder und Anwendungsverfahren
US4991214A (en) * 1987-08-28 1991-02-05 British Telecommunications Public Limited Company Speech coding using sparse vector codebook and cyclic shift techniques
US5010574A (en) * 1989-06-13 1991-04-23 At&T Bell Laboratories Vector quantizer search arrangement
US4975956A (en) * 1989-07-26 1990-12-04 Itt Corporation Low-bit-rate speech coder using LPC data reduction processing
US5305332A (en) * 1990-05-28 1994-04-19 Nec Corporation Speech decoder for high quality reproduced speech through interpolation
US5230037A (en) * 1990-10-16 1993-07-20 International Business Machines Corporation Phonetic hidden markov model speech synthesizer
US5396576A (en) * 1991-05-22 1995-03-07 Nippon Telegraph And Telephone Corporation Speech coding and decoding methods using adaptive and random code books
US5432883A (en) * 1992-04-24 1995-07-11 Olympus Optical Co., Ltd. Voice coding apparatus with synthesized speech LPC code book
US5321793A (en) * 1992-07-31 1994-06-14 SIP--Societa Italiana per l'Esercizio delle Telecommunicazioni P.A. Low-delay audio signal coder, using analysis-by-synthesis techniques
US5426460A (en) * 1993-12-17 1995-06-20 At&T Corp. Virtual multimedia service for mass market connectivity

Non-Patent Citations (14)

* Cited by examiner, † Cited by third party
Title
E.M. Warrington et al., "A Case Study on Digital Communication Systems," in. N.B. Jones et al., ed., Digital Signal Processing: Principles, Devices, and Applications, 1990, pp. 335-337.
E.M. Warrington et al., A Case Study on Digital Communication Systems, in. N.B. Jones et al., ed., Digital Signal Processing: Principles, Devices, and Applications, 1990, pp. 335 337. *
Hagen et al., "Low Bit-Rate Spectral Coding in CELP, A New LSP-Method," ICASSP-90, Apr. 3-6, 1990, v. 1, pp. 189-219.
Hagen et al., Low Bit Rate Spectral Coding in CELP, A New LSP Method, ICASSP 90, Apr. 3 6, 1990, v. 1, pp. 189 219. *
J. H. Chen et al., LD CELP: A High Quality 16 kb/s Speech Coder with Low Delay, IEEE Global Telecommunications Conf.: GLOBECOM 90, pp. 528 532 (1990). *
J. H. Chen, High Quality 16 kb/s Speech Coding with a One Way Delay Less Than 2 ms, Acoustics, Speech & Signal Processing Conference: ICASSP 90, pp. 453 456 (1990). *
J.-H. Chen et al., "LD-CELP: A High Quality 16 kb/s Speech Coder with Low Delay," IEEE Global Telecommunications Conf.: GLOBECOM '90, pp. 528-532 (1990).
J.-H. Chen, "High-Quality 16 kb/s Speech Coding with a One-Way Delay Less Than 2 ms," Acoustics, Speech & Signal Processing Conference: ICASSP '90, pp. 453-456 (1990).
Kuo et al., "Low Bit-Rate Quantization of LSP Parameters Using Two-Dimensional Differential Coding," ICASSP-92, Mar. 23-26, 1992, v. 1, pp. 97-100.
Kuo et al., Low Bit Rate Quantization of LSP Parameters Using Two Dimensional Differential Coding, ICASSP 92, Mar. 23 26, 1992, v. 1, pp. 97 100. *
Ozawa et al., "4kb/s Improved CELP Coder with Efficient Vector Quantization," ICASSP-91, May 14-17, 1991, v. 1, pp. 213-216.
Ozawa et al., 4kb/s Improved CELP Coder with Efficient Vector Quantization, ICASSP 91, May 14 17, 1991, v. 1, pp. 213 216. *
Xydeas et al., "A Long History Quantization Approach to Scalar and Vector Quantization of LSP Coefficients," ICASSP-93, Apr. 27-30, 1993, v. 2, pp. 1-4.
Xydeas et al., A Long History Quantization Approach to Scalar and Vector Quantization of LSP Coefficients, ICASSP 93, Apr. 27 30, 1993, v. 2, pp. 1 4. *

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US5909663A (en) * 1996-09-18 1999-06-01 Sony Corporation Speech decoding method and apparatus for selecting random noise codevectors as excitation signals for an unvoiced speech frame
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US20020055836A1 (en) * 1997-01-27 2002-05-09 Toshiyuki Nomura Speech coder/decoder
US6073092A (en) * 1997-06-26 2000-06-06 Telogy Networks, Inc. Method for speech coding based on a code excited linear prediction (CELP) model
US6799161B2 (en) * 1998-06-19 2004-09-28 Oki Electric Industry Co., Ltd. Variable bit rate speech encoding after gain suppression
US20030105624A1 (en) * 1998-06-19 2003-06-05 Oki Electric Industry Co., Ltd. Speech coding apparatus
WO2000008633A1 (fr) * 1998-08-06 2000-02-17 Matsushita Electric Industrial Co., Ltd. Generateur de signaux d'excitation, codeur vocal et decodeur vocal
US20020103638A1 (en) * 1998-08-24 2002-08-01 Conexant System, Inc System for improved use of pitch enhancement with subcodebooks
US7117146B2 (en) * 1998-08-24 2006-10-03 Mindspeed Technologies, Inc. System for improved use of pitch enhancement with subcodebooks
US6138089A (en) * 1999-03-10 2000-10-24 Infolio, Inc. Apparatus system and method for speech compression and decompression
US6594626B2 (en) * 1999-09-14 2003-07-15 Fujitsu Limited Voice encoding and voice decoding using an adaptive codebook and an algebraic codebook
US6738733B1 (en) * 1999-09-30 2004-05-18 Stmicroelectronics Asia Pacific Pte Ltd. G.723.1 audio encoder
US20080312917A1 (en) * 2000-04-24 2008-12-18 Qualcomm Incorporated Method and apparatus for predictively quantizing voiced speech
US8660840B2 (en) * 2000-04-24 2014-02-25 Qualcomm Incorporated Method and apparatus for predictively quantizing voiced speech
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US7114072B2 (en) * 2000-12-30 2006-09-26 Electronics And Telecommunications Research Institute Apparatus and method for watermark embedding and detection using linear prediction analysis
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US20040093207A1 (en) * 2002-11-08 2004-05-13 Ashley James P. Method and apparatus for coding an informational signal
US7054807B2 (en) * 2002-11-08 2006-05-30 Motorola, Inc. Optimizing encoder for efficiently determining analysis-by-synthesis codebook-related parameters
US20090024395A1 (en) * 2004-01-19 2009-01-22 Matsushita Electric Industrial Co., Ltd. Audio signal encoding method, audio signal decoding method, transmitter, receiver, and wireless microphone system
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